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A Feasibility Study of Powerline Communication Technology for Digital Inclusion In Brazilian Amazon (Proceedings Paper)

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A Feasibility Study of Powerline Communication Technology for Digital Inclusion In Brazilian Amazon (Proceedings Paper)
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    A feasibility study of powerline communication technology for digitalinclusion in Brazilian Amazon Jorge A. M. de Souza *a , Marcelino S. da Silva a , Carlos R. L. Francês a ,João C. W. A. Costa a , Ádamo Santana a , Marcelo E. V. Segatto  b , Flavio R. Antonio c ,Gabryella Rodrigues a   a Federal University of Pará, 01 Augusto Corrêa St., Belém, PA, BRAZIL 66075-900;  b Federal University of Espírito Santo, 514 Fernando Ferrari Ave, Vitória, ES, BRAZIL 29075-910; c Eletronorte, SQN Qd. 06 – Cj. A Bl. B and C, Brasília, DF, BRAZIL 70718-900 ABSTRACT In the current national scene, many actions point at projects of digital inclusion and citizenship. In this context, providingaccess technologies as a requisite for the implementation of these actions is primordial. In this way, many innovativeexperiences have been presented in the past few years. This paper presents a study on the Powerline Communication – PLC technology; as a proposal for a feasible access network for Brazilian Amazon. First, the characteristics of the PLCtechnology are studied from an implanted indoor prototype at Federal University of Pará. The measures used in this prototype serve as input for a created model, from which it is intended to study the system more widely, consideringfactors such as: scalability, reliability and the physical characteristics. Keywords: Power line communication, VoIP, digital inclusion 1.   INTRODUCTION The Brazilian Amazon has particular characteristics that motivate a great number of researches in all knowledge areasand general interest of the entire world. Despite the natural appeal, the under development of the region promotes manyactions to diminish the existent problems. One of then is the poor communication infrastructure, not only last milecommunication, but even indoor, which increases the digital exclusion. The Federal University of Pará, particularly, theHigh Performance Networks Planning Laboratory, has being conducting its efforts on communication technologiesresearches (e.g. Digital Subscriber Line - DSL and Powerline Communication - PLC) with an approach strongly directedto the peculiar characteristics of the Amazon region, toward the digital inclusion, as [1] and [2].In this context, actions such teleeducation stands out due to joint use of technology and education, contributing to digitaland social inclusion of people. An example is the possibility of training a group of people, in different classrooms, in thesame time, with only one teacher. This paper presents a feasibility study of powerline communication technology appliedto digital inclusion with a case study of VoIP.There have been many researches on VoIP over consolidated technologies. In [15] was made a performance evaluationof VoIP over 3G-WLANInternetworking system focused in the performance of VoIP with IPSec tunnel in 3G-WLANinternetworking system, the maximum number of VoIP connections in a single 802.11b Access Point, the handoff latency due to terminal mobility and the impact of terminal mobility on the VoIP performance. In [16] the interest is inverify the QoS in terms of MOS value on VoIP over WLANS. [17] presents a performance evaluation of VoIP over satellite networks under different link and traffic conditions based on laboratory experiments. [18] study the performanceof VoIP over GPRS in terms of capacity of voice connections compared with a traditional circuit switch.The characteristics of PLC technology are shown in section 2. Section 3 shows the main characteristics of VoIP, whichhave a straight influence on quality of the transmitted voice. The performance evaluation of VoIP over PLC is presentedin section 4 and a case study in section 5. * jamsouza@ufpa.br; phone +55 03191 3201 8112; fax +55 03191 3201 7740; www.lprad.ufpa.br   Proceedings of SPIE Optics East, v. 6390, p. 63900I, 2006     2.   POWERLINE COMMUNICATION The PLC technology uses the existing electric infrastructure to provide the data communication. This technology hasshown itself sufficiently competitive in the broadband access market, disputing with other more traditional, like DSL.PLC counts with the advantage of having a sufficiently wide infrastructure and with relatively little cost associated [3].PLC, however, as well as other data communication technologies, possesses some inconveniences. The physicalenvironment is very hostile for the data, once that it was not conceived for this purpose, thus there are many properties of the power systems which influence negatively the high speed communications (losses in the cable, propagation inmultiple paths and the noise, for example) [4].A way to reduce the impact of the transmission environment in the communication is the application of efficientmodulation methods, such as OFDM (Ortogonal Frequency Division Multiplexing), besides mechanisms for error correction such as FEC (Forward Error Correction) and the ARQ (Automatic Repeat reQuest) [5].However, independent of the adopted techniques, a factor to be observed and understood is the noise in thecommunication. 2.1   Noise Characteristics Given the physical characteristics of the conductor and the nature of the electric energy, this environment is not bestindicated for data communication. The propagation of the signal through the power transmission line provokes anattenuation and a delay in the signal, which increases with the distance and the frequency. This attenuation depends onthe impedance characteristic (Z L ) and on the propagation constant ( γ ) of the transmission lines. In accordance with [13]and [1], these two parameters are resistance functions R, conductance G, inductance L, and capacitance C, per unit of length, which depend on the frequency.Based on the studies presented in [6] and [7], two types of noises can be adopted: (a) deep noise, formed by deep colorednoises, short band noises and asynchronous periodic impulsive noises in relation to the channel frequency; and (b)impulsive noise, formed by asynchronous impulsive noises and synchronous periodic impulsive noises in relation to thechannel frequency. Having the latter a great probability to cause bit errors or in bits sequence in a of high speed datacommunication [4].With the purpose to characterize the typical traffic, as well as the essential characteristics of this type of communicationregarding the communication channel and the noises, a PLC prototype, which will be described in section 3, is beingtested. 2.2   PLC channel model The topology of the power distribution networks considerably differs from the traditional communication networks, suchas twisted pair, coaxial cable or optic fiber. Numerous reflections of the signal are received and occur mainly due the junction of cables of different impedances.The PLC channel has the characteristic of being selective to the frequency, this if due to the multipath characteristic of the power transmission networks.Given the hostile characteristics of PLC channel (multipath, selective to the frequency and the impulsive noise) it isnecessary to apply an efficient modulation technique.For allowing the transmission of data through many independent sub-carriers, the OFDM modulation technique has beena good alternative for the data transmission on the PLC channel. Once that only a small percentage of sub-carriers will bereached by the effect of frequency selectivity of the canal, error correction codes can be used to correct the data of thefew wrong sub-carriers, as well as new sub-carriers can be used to substitute the wrong ones.The OFDM modulation characteristics regarding the impulsive noise and the multipath are presented in details in [8].  Proceedings of SPIE Optics East, v. 6390, p. 63900I, 2006     3.   VOIP The use of voice over IP (VoIP) comes being one of the great goals of investments for solution suppliers and users of telecommunication in last and next years. As the name implies, VoIP refers to calls that traverse networks using InternetProtocol (IP). This technology came to change the way we communicate on the Web, as well as the way we do business,it opens a new horizon for combining voice and data in the same terminal equipment of user, increasing the number of applications to be accessible through the same media for transmission and diminishing the costs of communication whencompared with the conventional interurban calls.VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks, like many VoIP packages include PSTN features that most telcos normally charge extra for, or maybe be unavailable, such as integrationof data, voice and fax, sound grading, video telephony, unified messaging and web-based call centers, which are notavailable in traditional telephone networks [9]. It also allows users to travel anywhere in the world and the incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network.The first process in an IP voice system is the digitization of the speaker’s voice. Then occurs the compression of thesignal, after that the system examines the digitalized data to discards any packets that do not contains a speech. Second,some algorithms and sophisticated codecs are employed to reduce the amount of information that must be sent andcompression the voice streams respectively. Just remembering that a normal telephone uses circuit switching for phonecalls, which involves routing of your call through the switch at your local carrier to the person you are calling. Packetswitching on the other hand is more efficient in transmitting data since a packet is sent from one system to another,having several advantages over circuit switching. The key difference between packet and circuit switching is that the firstone allows numerous telephone calls to inhabit the space occupied by only one in a circuit-switched network.VoIP is best understood as a collection of the protocols that make up its mechanics. Those protocols are looselyanalogous to the PSTN (Public Switched Telephone Network), which is broken down into three categories: access,switching, and transport. Three categories of protocols are relevant to VoIP: signaling, routing and transporting. 3.1   H.323 Protocol The H.323 is an ITU-T standard [10] that does not guarantee QoS and it was developed before the emergence of VoIP, but it is the most widely used VoIP standard. This protocol the base technology for the transmission of real-time audio,video and data communication over packet based networks. It specifies the components, protocols and procedures that provide multimedia communication over packet based networks (include LANs and WANs).It addresses call control, multimedia management, bandwidth management and interfaces between LANs and other network. The entities in the H.323 topology are the endpoints, also called user terminals, gateways (GW), multipointcontrol units (MCU) and gatekeeper (GK).The gateway perform the translation of signaling and media exchange between H.323 and PSTN endpoint [11], for example, a VoIP gateway provides translation of transmission formats and signaling procedures between a telephoneswitched circuit gateway network (SCN) and a packet network. In addition, the VoIP gateway may perform speechtranscoding and compression, and it is usually capable of generating and detecting DTMF signals [12].Although MCUs, though listed separately, are in practice part of a gatekeeper or a high speed computer that acts as aterminal serving one or more users. And the gatekeeper, that is used for admission control, address translation function,manage network bandwidth, allocate voice, video and data, handle routing for multiple H323 gateways, provide for charging and billing and support network security and subscriber authentication.On the other hand to set up a call using the H.323 is really complexity and takes a long time. First, the protocol usesmultiple roundtrip messages to establish signaling and control for any call between two terminals. Moreover, H.323requires that TCP connections be used to carry the messages, requiring an additional roundtrip exchange. The recentlyreleased version 3 is an improvement and includes both a “fast connect” procedure that effectively consolidates theQ.931 messages exchange between terminals, and a tunneling procedure that lets H.245 share a single TCP connectionwith Q.931 [13]. 3.2   Quality of Service on VoIP applications When packets are lost and delayed in any point of the transmission through the network, VoIP users will see a drop-outof voice.  Proceedings of SPIE Optics East, v. 6390, p. 63900I, 2006     What is more noticeable in case of congestion, that some packets could be lost or delayed in the network (one of major challenges is the provision of Quality of Service, currently IP does not offer any quality of QoS guarantees to thetransmitted data), also a percentage of packet are discarded due to errors that occurs during the transmission. These problems results in a substantial deterioration of voice quality. Because VoIP systems are time sensitive and cannot waitfor retransmission, more sophisticated error detection and correction systems are used to create sound to fill in the gaps[9].So, there are some things to consider:• Latency: Delay for packet delivery;• Jitter: Variations in delay of packet delivery;• Packet loss: Too much traffic in the network causes the network to drop packets;• Burstiness of Loss and Jitter: Loss and Discards (due to jitter) tend to occur in bursts;Jitter causes strange sound effects, but can be handled to some degree with "jitter buffers" in the software. Packet losscauses interrupts. Some degree of packet loss won't be noticeable, but lots of packet loss will make sound lousy.To correct the packet loss it can be used some codecs algorithms, but they are effective only if a single packet is lostduring the period of transmission. Today there are many codecs available for digitizing speech, such as ITU-T G.711(PCM), G.723 (MP-MLQ), G.726 (ADPCM), G.728 (LD-CELP) and G.729 (CS-ACELP).A very important factor affecting voice quality is the total network load, what was said previously. When the congestionin the network is high, jitter and frame loss will increase. So to get measurements of the network load, as well as thenumber of collisions, many different tools are available (protocols analyzers). To gauge the effect of priorities, it is possible to use the latency and loss application to ensure that priorities are configured correctly and indeed voice is given precedence over data traffic.Another factor used to determine the quality of transmitted speech is the benchmark called Mean Opinion Score (MOS).With MOS, a number of listeners judge the quality of a voice sample on a scale of 1 to 5, where 1 is lowest perceivedquality, and 5 is the highest perceived quality [16]. The average of the scores provides the MOS for that sample. Table 1shows the rating scheme used by listeners to judge the voice sample. Each codec implementation has a Mean OpinionScore, the MOS of G.711, codec used in the experiments, is 4.1. Table 1. Mean Opinion Score (MOS) MOS Quality Impairment 5 Excellent Imperceptible4 Good Perceptible but not annoying3 Fair Slightly annoying2 Poor Annoying1 Bad Very annoying 4.   PERFORMANCE EVALUATION To perform the study of VoIP over PLC a PLC prototype was used in conjunction with RADCOM© Performer Analyzer, which has a VoIP module that gives values such as jitter, packet loss and MOS. In sub-section 4.1 the testbedis presented. 4.1   Testbed of VoIP over PLC The PLC Indoor (PLC/LPRAD) prototype is located in the dependences of the Laboratory of Electric Engineering andComputing - LEEC of the UFPA, in which is inserted the Laboratory of High Performance Networks Planning - LPRAD- and has the purpose of supplying information regarding the nature of the noise (physical layer), supply typicalinformation the superior layers, load characterization and performance measures of the system (e.g. trustworthiness and  Proceedings of SPIE Optics East, v. 6390, p. 63900I, 2006     scalability). The information obtained in this prototype is serving as entry parameters for the simulations of PLC network in the NS-2 [14].The sub-station of the LEEC, after the transforming exit from medium to low voltage, possess two circuits, one for the building illumination, feeding of the air conditioners and for some outlets in building walls, and the other, for the power keys located in the building rooms, from which each room mounts its internal power system.In this sub-station a PLC base station was installed (HE), which is connected to the three phases of the circuit that feedsthe power keys of the rooms; this way, all the rooms of the LEEC will possess internet connectivity points through thePLC network. This circuit was chosen given the fact of being wider than the first one.In LPRAD, there is a PLC MODEM with a machine that is the transmitter, and in another lab of LEEC there is a set PLCMODEM/ RADCOM© Performer Analyzer. The Figure 1 shows the VoIP over PLC prototype. Fig. 1. VoIP over PLC Prototype Topology. 4.2   VoIP over PLC experiment The experiment was performed using the prototype, previous described, with four remote nodes, used one by one,located in different points of the LEEC, two in the first floor (Power System Laboratory and Digital ElectronicsLaboratory) and two in the second floor (LPRAD and Laboratory of Applied Electromagnetism - LEA). To generate theVoIP calls, a software called CallGen323 was used, this software was developed by the OpenH232 Project and isavailable in [21]. The main objective of the OpenH323 Project is “to create a full featured, interoperable, open sourceimplementation of the ITU-T H.323 teleconferencing protocol that can be used by personal developers and commercialusers without charge.” Fig. 2. LEEC Building Each VoIP call had duration of three minutes, repeated twenty times, and total audio bytes of 81% and silencesuppression of 20.38%. Each experiment takes one hour to be completed. The voice file used was obtained in [19]. Allthe metrics relative to VoIP calls were captured by MediaPro, a software module of RADCOM© Performer Analyzer.More information about MediaPro and RADCOM© Performer Analyzer can be found in [20].  Proceedings of SPIE Optics East, v. 6390, p. 63900I, 2006 
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